Movatterモバイル変換


[0]ホーム

URL:


US10896668B2 - Signal processing apparatus, signal processing method, and computer program - Google Patents

Signal processing apparatus, signal processing method, and computer program
Download PDF

Info

Publication number
US10896668B2
US10896668B2US16/480,381US201716480381AUS10896668B2US 10896668 B2US10896668 B2US 10896668B2US 201716480381 AUS201716480381 AUS 201716480381AUS 10896668 B2US10896668 B2US 10896668B2
Authority
US
United States
Prior art keywords
digital signal
bit number
signal
quantization bit
sampling frequency
Prior art date
Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
Active
Application number
US16/480,381
Other versions
US20190385585A1 (en
Inventor
Yoshinori Tamori
Kohei Asada
Tetsunori Itabashi
Shinpei Tsuchiya
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
Sony Corp
Original Assignee
Sony Corp
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed by Sony CorpfiledCriticalSony Corp
Assigned to SONY CORPORATIONreassignmentSONY CORPORATIONASSIGNMENT OF ASSIGNORS INTEREST (SEE DOCUMENT FOR DETAILS).Assignors: ITABASHI, TETSUNORI, ASADA, KOHEI, TAMORI, Yoshinori, TSUCHIYA, Shinpei
Publication of US20190385585A1publicationCriticalpatent/US20190385585A1/en
Application grantedgrantedCritical
Publication of US10896668B2publicationCriticalpatent/US10896668B2/en
Activelegal-statusCriticalCurrent
Anticipated expirationlegal-statusCritical

Links

Images

Classifications

Definitions

Landscapes

Abstract

A signal processing apparatus is provided which can suppress external noise without degrading an audio characteristic.
Provided is a signal processing apparatus including: an A/D converter configured to output a digital signal having a predetermined sampling frequency and quantization bit number a, the A/D converter including a first delta sigma modulator that performs a first delta sigma modulation process on an input analog signal; a filter unit configured to pass an output of the A/D converter through a digital filter provided with a predetermined filter characteristic and output a digital signal having the sampling frequency and a quantization bit number b; a second delta sigma modulator configured to perform a second delta sigma modulation process on an output of the filter unit and output a digital signal having the sampling frequency and the quantization bit number a; and an addition unit configured to add an output of the second delta sigma modulator and an input digital signal having the sampling frequency and the quantization bit number.

Description

CROSS-REFERENCE TO RELATED APPLICATIONS
This application claims the benefit under 35 U.S.C. § 371 as a U.S. National Stage Entry of International Application No. PCT/JP2017/044374, filed in the Japanese Patent Office as a Receiving Office on Dec. 11, 2017, which claims priority to Japanese Patent Application Number JP 2017-015807, filed in the Japanese Patent Office on Jan. 31, 2017, each of which is hereby incorporated by reference in its entirety.
TECHNICAL FIELD
The present disclosure relates to a signal processing apparatus, a signal processing method, and a computer program.
BACKGROUND ART
A noise canceling system has been put into practical use to suppress external noise and enhance a sound insulation effect in a case where a listener listens to audio content with an audio reproduction device such as headphones and earphones. A general noise canceling system generates a signal that cancels noise collected by a noise detection microphone and adds the signal to an audio signal to suppress external noise (SeePatent Documents 1, 2, and the like).
CITATION LISTPatent Documents
Patent Document 1: Japanese Patent Application Laid-Open No. 2008-193421
Patent Document 2: Japanese Patent Application Laid-Open No. 2009-33309
SUMMARY OF THE INVENTIONProblems to be Solved by the Invention
In a case where an audio signal in a direct stream digital (DSD) system having a sampling frequency of megahertz (for example, 2.8 MHz) and a quantization bit number of one is used as an audio signal in an existing noise canceling system, the addition of quantization noise causes degradation of the audio characteristic.
Thus, the present disclosure proposes a signal processing apparatus, a signal processing method, and a computer program that are new and improved and can suppress external noise without degrading an audio characteristic.
Solutions to Problems
The present disclosure provides a signal processing apparatus including: an A/D converter configured to output a digital signal having a predetermined sampling frequency and quantization bit number a, the A/D converter including a first delta sigma modulator that performs a first delta sigma modulation process on an input analog signal; a filter unit configured to pass an output of the A/D converter through a digital filter provided with a predetermined filter characteristic and output a digital signal having the sampling frequency and a quantization bit number b; a second delta sigma modulator configured to perform a second delta sigma modulation process on an output of the filter unit and output a digital signal having the sampling frequency and the quantization bit number a; and an addition unit configured to add an output of the second delta sigma modulator and an input digital signal having the sampling frequency and the quantization bit number a.
Furthermore, the present disclosure provides a signal processing apparatus including: an A/D converter configured to output a digital signal having a predetermined sampling frequency and a quantization bit number a, the A/D converter including a first delta sigma modulator that performs a first delta sigma modulation process on an input analog signal; a filter unit configured to pass an output of the A/D converter through a digital filter provided with a predetermined filter characteristic and output a digital signal having the sampling frequency and a quantization bit number b; a second delta sigma modulator configured to perform a second delta sigma modulation process on an output of the filter unit and output a digital signal having the sampling frequency and the quantization bit number a; a first bit expander configured to expand the quantization bit number from a to c for an output of the second delta sigma modulator; and a first addition unit configured to add an output of the first bit expander and an input digital signal having the sampling frequency and a quantization bit number c.
Further, the present disclosure provides a signal processing apparatus including: a first delta sigma modulation unit configured to perform a first delta sigma modulation process on an input analog signal, generate a digital signal having a predetermined sampling frequency and a quantization bit number a, and output the digital signal after expanding its quantization bit number from a to c; a first equalizer unit configured to generate a first equalized signal by equalizing an input digital signal having the sampling frequency and a quantization bit number c with a first target characteristic, perform a delta sigma modulation process on the first equalized signal, generate a digital signal having the predetermined sampling frequency and the quantization bit number a, and output the digital signal after expanding its quantization bit number from a to c; a second equalizer unit configured to generate a second equalized signal by equalizing an input digital signal having the sampling frequency and the quantization bit number c with a second target characteristic, perform the delta sigma modulation process on the second equalized signal, generate a digital signal having the predetermined sampling frequency and the quantization bit number a, and output the digital signal after expanding its quantization bit number from a to c; a delay unit configured to provide the input digital signal with a delay equivalent to processing delay in the first equalizer unit or the second equalizer unit, and output the signal after expanding its quantization bit number from a to c; a first addition unit configured to add outputs of the first delta sigma modulation unit, the delay unit, and the first equalizer unit; a filter unit configured to pass an output of the first addition unit through a digital filter provided with a predetermined filter characteristic and output a digital signal having the sampling frequency and a quantization bit number b; a second delta sigma modulation unit configured to perform a delta sigma modulation process on an output of the filter unit, generate a digital signal having the sampling frequency and the quantization bit number a, and output the digital signal after expanding its quantization bit number from a to c; and a second addition unit configured to add outputs of the second delta sigma modulation unit, the delay unit, and the second equalizer unit.
Furthermore, the present disclosure provides a signal processing method including: outputting a digital signal having a predetermined sampling frequency and a quantization bit number a by performing a first delta sigma modulation process on an input analog signal; passing a digital signal having the predetermined sampling frequency and the quantization bit number a through a digital filter provided with a predetermined filter characteristic and outputting a digital signal having the sampling frequency and a quantization bit number b; outputting a digital signal having the sampling frequency and the quantization bit number a by performing a second delta sigma modulation process on the digital signal having the quantization bit number b; and adding an output of the second delta sigma modulation process and an input digital signal having the sampling frequency and the quantization bit number a.
Further, the present disclosure provides a signal processing method including: outputting a digital signal having a predetermined sampling frequency and a quantization bit number a by performing a first delta sigma modulation process on an input analog signal; passing a digital signal having the predetermined sampling frequency and the quantization bit number a through a digital filter provided with a predetermined filter characteristic and outputting a digital signal having the sampling frequency and a quantization bit number b; outputting a digital signal having the sampling frequency and the quantization bit number a by performing a second delta sigma modulation process on an output of the digital signal having the sampling frequency and the quantization bit number b; expanding the quantization bit number from a to c for an output of the second delta sigma modulation process; and adding the digital signal having an expanded quantization bit number c and an input digital signal having the sampling frequency and a quantization bit number c.
Furthermore, the present disclosure provides a signal processing method including: performing a first delta sigma modulation process where a digital signal having a predetermined sampling frequency and a quantization bit number a is generated by performing a delta sigma modulation process on an input analog signal, and the digital signal is output after expanding its quantization bit number from a to c; performing a first equalization process where a first equalized signal is generated by equalizing an input digital signal having the sampling frequency and a quantization bit number c with a first target characteristic, a delta sigma modulation process is performed on the first equalized signal, a digital signal having the predetermined sampling frequency and the quantization bit number a is generated, and the digital signal is output after expanding its quantization bit number from a to c; performing a second equalization process where a second equalized signal is generated by equalizing an input digital signal having the sampling frequency and a quantization bit number c with a second target characteristic, a delta sigma modulation process is performed on the second equalized signal, a digital signal having the predetermined sampling frequency and the quantization bit number a is generated, and the digital signal is output after expanding its quantization bit number from a to c; performing a delay process where the input digital signal is provided with a delay equivalent to processing delay in the first equalization process or the second equalization process, and the signal is output after expanding its quantization bit number from a to c; performing a first addition process where outputs of the first delta sigma modulation process, the delay process, and the first equalization process; performing a filter process where an output of the first addition process is passed through a digital filter provided with a predetermined filter characteristic and, a digital signal having the sampling frequency and a quantization bit number b is output; performing a second delta sigma modulation process where a digital signal having the sampling frequency and the quantization bit number a is generated by performing a delta sigma modulation process on an output of the filter process and the digital signal is output after expanding its quantization bit number from a to c; and performing a first addition process where outputs of the second delta sigma modulation process, the delay process, and the second equalization process.
Furthermore, the present disclosure provides a computer program that causes a computer to execute: outputting a digital signal having a predetermined sampling frequency and a quantization bit number a by performing a first delta sigma modulation process on an input analog signal; passing a digital signal having the predetermined sampling frequency and the quantization bit number a through a digital filter provided with a predetermined filter characteristic and outputting a digital signal having the sampling frequency and a quantization bit number b; outputting a digital signal having the sampling frequency and the quantization bit number a by performing a second delta sigma modulation process on the digital signal having the quantization bit number b; and adding an output of the second delta sigma modulation process and an input digital signal having the sampling frequency and the quantization bit number a.
Furthermore, the present disclosure provides a computer program that causes a computer to execute: outputting a digital signal having a predetermined sampling frequency and a quantization bit number a by performing a first delta sigma modulation process on an input analog signal; passing a digital signal having the predetermined sampling frequency and the quantization bit number a through a digital filter provided with a predetermined filter characteristic and outputting a digital signal having the sampling frequency and a quantization bit number b; outputting a digital signal having the sampling frequency and the quantization bit number a by performing a second delta sigma modulation process on an output of the digital signal having the sampling frequency and the quantization bit number b; expanding the quantization bit number from a to c for an output of the second delta sigma modulation process; and adding the digital signal having an expanded quantization bit number c and an input digital signal having the sampling frequency and a quantization bit number c.
Furthermore, the present disclosure provides a computer program that causes a computer to execute: performing a first delta sigma modulation process where a digital signal having a predetermined sampling frequency and a quantization bit number a is generated by performing a delta sigma modulation process on an input analog signal, and the digital signal is output after expanding its quantization bit number from a to c; performing a first equalization process where a first equalized signal is generated by equalizing an input digital signal having the sampling frequency and a quantization bit number c with a first target characteristic, a delta sigma modulation process is performed on the first equalized signal, a digital signal having the predetermined sampling frequency and the quantization bit number a is generated, and the digital signal is output after expanding its quantization bit number from a to c; performing a second equalization process where a second equalized signal is generated by equalizing an input digital signal having the sampling frequency and a quantization bit number c with a second target characteristic, a delta sigma modulation process is performed on the second equalized signal, a digital signal having the predetermined sampling frequency and the quantization bit number a is generated, and the digital signal is output after expanding its quantization bit number from a to c; performing a delay process where the input digital signal is provided with a delay equivalent to processing delay in the first equalization process or the second equalization process, and the signal is output after expanding its quantization bit number from a to c; performing a first addition process where outputs of the first delta sigma modulation process, the delay process, and the first equalization process; performing a filter process where an output of the first addition process is passed through a digital filter provided with a predetermined filter characteristic, and a digital signal having the sampling frequency and a quantization bit number b is output; performing a second delta sigma modulation process where a digital signal having the sampling frequency and the quantization bit number a is generated by performing a delta sigma modulation process on an output of the filter process and the digital signal is output after expanding its quantization bit number from a to c; and performing a first addition process where outputs of the second delta sigma modulation process, the delay process, and the second equalization process.
Effects of the Invention
As described above, according to the present disclosure, a signal processing apparatus, a signal processing method, and a computer program which are new and improved and can suppress external noise without degrading an audio characteristic.
Here, the above described effect should not be limited, and there may be any one of the effects described in this specification or other effects that can be generated based on the present specification in addition to the above mentioned effects, together with the above mentioned effects, or as a substitute for the above mentioned effects.
BRIEF DESCRIPTION OF DRAWINGS
FIG. 1 is a diagram illustrating a configuration example of a noise canceling system according to a first embodiment of the present disclosure.
FIG. 2 is a diagram illustrating an example of the configuration of the noise canceling system according to a second embodiment of the present disclosure.
FIG. 3 is a diagram illustrating an example of the configuration of the noise canceling system according to a third embodiment of the present disclosure.
FIG. 4 is a diagram illustrating an example of the configuration of the noise canceling system according to a fourth embodiment of the present disclosure.
FIG. 5 is a diagram illustrating an example of the configuration of the noise canceling system according to a fifth embodiment of the present disclosure.
MODE FOR CARRYING OUT THE INVENTION
Preferred embodiments of the present disclosure will be described in detail below with reference to the accompanying drawings. Here, in the present specification and the drawings, same reference numerals are given to constituent elements having substantially same functional configuration, and redundant explanation will be omitted.
Note that the description will be given in the following order.
1. Overview
2. First embodiment (Feed-forward method)
3. Second embodiment (Feed-forward method)
4. Third embodiment (Feedback method)
5. Fourth embodiment (Feedback method and feed-forward method)
6. Fifth embodiment (Feedback method)
7. Summary
1. Overview
Before describing the embodiments of the present disclosure in detail, an overview of the embodiments of the present disclosure will be provided.
As described above, a general noise canceling system generates a signal that cancels noise collected by the noise detection microphone, and suppresses external noise by adding the signal to an audio signal. Then, as the noise canceling system, there are a feed-forward method, a feedback method, and a method combining the feed forward method and the feedback method. The feed-forward method is a method of performing signal processing to cancel a sound signal (external noise) collected by a microphone provided outside a housing of a headphone. The feedback method is a method of performing signal processing to cancel a sound signal (internal noise) collected by a microphone provided inside a housing of a headphone.
For example,Patent Document 1 discloses a technique for a noise canceling system that suppresses external noise using a feed-forward method. In the feed-forward method, bit extension is performed by a bit expander on an audio signal from a digital audio source to combine the signal with a noise canceling signal that is used to cancel a sound signal collected by a microphone. This process is performed to set a quantization bit number of the noise canceling signal and a quantization bit number of the audio signal to be corresponding to each other. In a case where the noise canceling signal and the audio signal are combined, the signals are converted to an analog signal through a delta sigma modulator and a low pass filter, and the sound is mainly output from headphones and earphones through an amplifier.
Also in the feedback method, bit extension is performed by an equalizer on an audio signal from a digital audio source to combine the signal with a noise canceling signal that is used to cancel a sound signal collected by a microphone. In a case where the noise canceling signal and the audio signal are combined, the signals are converted to an analog signal through a delta sigma modulator and a low pass filter, and the sound is mainly output from headphones and earphones through an amplifier.
Here, in a case where an audio signal of a DSD system in which a sampling frequency is in an order of megahertz (for example, 2.8 MHz) and a quantization bit number is 1 bit is used as the audio signal, the quantization bit number of the audio signal is expanded to 16 bits to combine the audio signal with the noise canceling signal in which the quantization bit number is 16 bits, for example. Then, in a case where the noise canceling signal and the audio signal are combined, the delta sigma modulator converts the combined signal into a signal in which the quantization bit number is one bit. Here, focusing on a characteristic change of the DSD audio signal, an addition of quantization noise cannot be avoided since the signal passes through the delta sigma modulator, and this causes deterioration of the audio characteristic. Deterioration of the audio characteristic of a signal that passes through the delta sigma modulator can be seen in the feedback method in a similar manner.
Therefore, in view of the above-mentioned issue, a person who discloses the present disclosure has eagerly studied on a noise canceling system capable of suppressing external noise without degrading audio characteristic of a digital audio source. As a result, the person of the present disclosure has invented a noise canceling system capable of suppressing external noise without degrading audio characteristic of a digital audio source, as described below.
The overview of the embodiments of the present disclosure has been described above.
2. First Embodiment (Feed-Forward Method)
As a first embodiment, an example of a feed-forward noise canceling system that does not degrade an audio characteristic of a digital audio source will be described.
FIG. 1 is a diagram illustrating a configuration example of a noise canceling system according to the first embodiment of the present disclosure. In the following, a configuration example of the noise canceling system according to the first embodiment of the present disclosure will be described with reference toFIG. 1.
As illustrated inFIG. 1, the noise canceling system according to the first embodiment of the present disclosure includes amicrophone111, anamplifier112, an A/D converter unit120, a noise cancelingdigital filter130, and adelta sigma modulator132, anadder134, a pulse width modulation (PWM)conversion unit136, an analog low pass filter (LPF)138, apower amplifier140, and aheadphone150. Theheadphone150 illustrated inFIG. 1 includesdrivers151 and152 compatible with a two-channel stereo of left (L) and right (R), but the configuration of the noise canceling system illustrated inFIG. 1 is compatible with at least one of an L channel or an R channel. Then, it is assumed that, in the noise canceling system illustrated inFIG. 1, the sampling frequency of the digital audio source is 64 Fs (2.8224 MHz), and the quantization bit number is one bit. Although the digital audio source in the noise canceling system illustrated inFIG. 1 is assumed to be a DSD audio source, the present disclosure is not limited to this example.
Themicrophone111 collects external sound (external noise), which is to be canceled in a vicinity of theheadphone150. In the feed-forward noise canceling system, themicrophones111 are respectively provided outside a housing of each of L and R single-side channels of theheadphone150 in actual. InFIG. 1, themicrophone111 provided corresponding to one of the L channel and the R channel is illustrated.
Theamplifier112 amplifies the external sound collected by themicrophone111 and an analog audio signal is obtained.
The A/D converter unit120 converts the analog audio signal output from theamplifier112 into a digital audio signal. The A/D converter unit120 includes adelta sigma modulator121. The delta sigma modulator121 converts the analog audio signal output from theamplifier112 into a digital signal of the same sampling frequency (64 Fs) and quantization bit number (one bit) as those of the digital audio source. Here, in the following description and the drawings, the sampling frequency of the signal and the quantization bit number are denoted as [sampling frequency, quantization bit number]. In a case where it is denoted as [64 Fs, 1 bit], the signal has the sampling frequency of 64 Fs and the quantization bit number of one bit.
The noise cancelingdigital filter130 receives the digital audio signal output from the A/D converter unit120, that is, the digital audio signal obtained by collecting the external sound collected by themicrophone111. Then, using the input digital audio signal, the noise cancelingdigital filter130 generates an audio signal (sound signal for cancellation) of sound that is effective to cancel external sound that can reach an ear of a wearer of theheadphone150 and be heard corresponding to thedriver151 as sound to be output from thedriver151. As the simplest sound signal for cancellation, for example, the audio signal input to the noise cancelingdigital filter130, that is, the audio signal obtained by collecting the external sound has an inverse characteristic and an antiphase. In actual, a characteristic considering the transfer characteristic such as circuits and spaces in the system of the noise canceling system are provided.
The noise cancelingdigital filter130 is configured as, for example, a finite impulse response (FIR) filter. In the present embodiment, the noise cancelingdigital filter130 is configured as a filter whose input is [64 Fs, 1 bit] and whose output is [64 Fs, 16 bits]. Therefore, the output of the noise cancelingdigital filter130 is converted into a multi-bit.
The delta sigma modulator132 converts the quantization bit number in the [64 Fs, 16 bits] digital signal output from the noise cancelingdigital filter130 into one bit. In other words, thedelta sigma modulator132 generates a [64 Fs, 1 bit] digital signal from the [64 Fs, 16 bits] digital signal output from the noise cancelingdigital filter130.
Theadder134 adds the signal of the digital audio source and the signal output from thedelta sigma modulator132. Regarding the signal after the addition by theadder134, since two signals that can take binary values of 0 and 1 are added, the signal becomes a 2-bit signal that can take three values of 0, 1, and 2. That is, theadder134 generates a [64 Fs, 2 bits] digital signal.
The noise cancelingdigital filter130, thedelta sigma modulator132, and theadder134 can be provided, for example, in a digital signal processor (DSP). This DSP may be provided, for example, as a single chip component.
ThePWM conversion unit136 performs PWM modulation on the [64 Fs, 2 bits] digital signal output from theadder134. Then, theanalog LPF138 inputs a signal output from thePWM conversion unit136 and generates an analog audio signal. The analog audio signal generated by theanalog LPF138 is input to thepower amplifier140. Thepower amplifier140 amplifies the input audio signal and uses its output to drive thedriver151 in theheadphone150, corresponding to one of the ears.
Here, it is focused on a signal path of the digital audio source in the noise canceling system illustrated inFIG. 1. The signal of the digital audio source does not pass through the delta sigma modulator which may cause quantization noise. In other words, the signal of the digital audio source is synthesized as it is being a [64 Fs, 1 bit] digital signal with a [64 Fs, 1 bit] sound signal for canceling, and converted into an analog audio signal through thePWM conversion unit136 and theanalog LPF138 without passing through the delta sigma modulator.
Therefore, the noise canceling system according to the first embodiment of the present disclosure favorably delivers sound of the digital audio source to a listener without degrading an audio characteristic of the digital audio source in a case where external noise is suppressed.
3. Second Embodiment (Feed-Forward Method)
As a second embodiment, an example of a feed-forward noise canceling system will be described which does not degrade an audio characteristic of a digital audio source, as in the first embodiment.
FIG. 2 is a diagram illustrating a configuration example of a noise canceling system according to the second embodiment of the present disclosure. In the following, a configuration example of the noise canceling system according to the second embodiment of the present disclosure will be described with reference toFIG. 2. Although the digital audio source in the noise canceling system illustrated inFIG. 2 is a DSD audio source, the present disclosure is not limited to this example.
As compared with the noise canceling system illustrated inFIG. 1, the noise canceling system illustrated inFIG. 2 does not combine a signal from the digital audio source with a canceling sound signal. Although an input system to thedriver151 in the noise canceling system illustrated inFIG. 1 is represented by one system, in actual, thedriver151 has two terminals of positive and negative, and in the example illustrated inFIG. 1, one of the terminals is grounded. On the other hand, in the noise canceling system illustrated inFIG. 2, an analog signal based on the canceling sound signal is input to one of the terminals (the − terminal in the example ofFIG. 2) of thedriver151 and an analog signal based on the digital audio source is input to the other terminal (the + terminal in the example ofFIG. 2). In other words, in the noise canceling system illustrated inFIG. 2, thedriver151 has a form of bridged transformer less (BTL) connection.
That is, in the noise canceling system illustrated inFIG. 2, PWM modulation is performed in thePWM conversion unit136.
In addition to the synthesis by theadder134 in the noise canceling system illustrated inFIG. 1, the BTL connection illustrated inFIG. 2 is capable of favorably delivering the sound of the digital audio source to the listener without degrading the audio characteristic of the digital audio source in a case where external noise is suppressed.
4. Third Embodiment (Feedback Method)
As a third embodiment, an example of a feedback noise canceling system will be described which does not degrade an audio characteristic of a digital audio source.
FIG. 3 is a diagram illustrating a configuration example of a noise canceling system according to the third embodiment of the present disclosure. In the following, a configuration example of the noise canceling system according to the third embodiment of the present disclosure will be described with reference toFIG. 3.
As illustrated inFIG. 3, the noise canceling system according to the first embodiment of the present disclosure includes a microphone211, anamplifier212, an A/D converter unit220, a noise cancelingdigital filter230, anddelta sigma modulators232 and243,bit expanders234,244, and245, anequalizer241, adelayer242,adders246 and247, aPWM conversion unit248, ananalog LPF249, apower amplifier250, and aheadphone260. Theheadphone260 illustrated inFIG. 3 includes drivers261 and262 and corresponds to 2-channel stereo with left (L) and right (R), but the configuration of the noise canceling system illustrated inFIG. 3 corresponds to at least one of an L channel or an R channel. Then, the digital audio source in the noise canceling system illustrated inFIG. 3 is assumed to be [64 Fs, 1 bit]. Although the digital audio source in the noise canceling system illustrated inFIG. 3 is a DSD audio source, the present disclosure is not limited to this example.
The microphone211 collects sound output from the driver261 and external sound intruding into an inside of the housing of theheadphone260 to be canceled. In the feedback noise canceling system, the microphone211 is actually provided inside the corresponding housing respectively for the L and R one-side channels of theheadphones260. InFIG. 3, it is assumed that microphone211 provided corresponding to one of the L channel and R channel is illustrated.
Theamplifier212 amplifies the external sound collected by the microphone211 and an analog audio signal is obtained.
The A/D converter unit220 converts the analog audio signal output from theamplifier212 into a digital audio signal. The A/D converter unit220 includes adelta sigma modulator221. The delta sigma modulator221 converts the analog audio signal output from theamplifier212 into a digital signal having same [64 Fs, 1 bit] as the digital audio source.
The noise cancelingdigital filter230 inputs a digital audio signal, which is obtained by collecting the digital audio signal output from the A/D converter unit220, that is, internal sound of the housing on a side of the driver261 of theheadphone260 collected by the microphone211. Then, the noise cancelingdigital filter230 uses the input digital audio signal and generates an audio signal (a sound signal for cancellation) of sound, which is effective to cancel external sound that can be heard by reaching an ear of the wearer of theheadphone260 corresponding to the driver261 as a sound to be output from the driver261. More specifically, the noise cancelingdigital filter230 processes to provide a predetermined transfer function −β for noise cancellation to the sound collected by the microphone211.
In the present embodiment, the noise cancelingdigital filter230 is configured as a filter whose input is [64 Fs, 1 bit] and whose output is [64 Fs, 16 bits]. Therefore, the output of the noise cancelingdigital filter230 is converted into a multi-bit.
The delta sigma modulator232 converts the quantization bit number in the [64 Fs, 16 bits] digital signal output from the noise cancelingdigital filter230 into one bit. In other words, thedelta sigma modulator232 generates a [64 Fs, 1 bit] digital signal from the [64 Fs, 16 bits] digital signal output from the noise cancelingdigital filter230.
The bit expander234 converts the [64 Fs, 1 bit] digital signal output from the delta sigma modulator232 into a [64 Fs, 3 bits] digital signal in this example. More specifically, thebit expander234 respectively converts a value of the signal to “001” (0.25) if the value of the signal is “1” and to “111” (−0.25) if the value is “0”.
Theequalizer241 provides the digital audio source with a characteristic based on the transfer function of the coefficient13. Here, theequalizer241 converts a [64 Fs, 1 bit] digital signal into a [64 Fs, 16 bits] digital signal in this example. Thedelta sigma modulator243 performs delta sigma modulation on the output of theequalizer241 and converts the output into a [64 Fs, 1 bit] digital signal. Thedelayer242 performs predetermined delay processing on the signal from the digital audio source in accordance with a delay due to the signal processing of theequalizer241 and thedelta sigma modulator243.
The bit expander244 converts the [64 Fs, 1 bit] digital signal output from the delta sigma modulator243 into a [64 Fs, 3 bits] digital signal in this example. More specifically, thebit expander244 respectively converts a value of the signal to “001” (0.25) if the value is “1” and to “111” (−0.25) if the value is “0”. Furthermore, thebit expander244 converts the [64 Fs, 1 bit] digital signal output from thedelayer242 into a [64 Fs, 3 bits] digital signal. More specifically, thebit expander245 respectively converts the signal value to “001” (0.25) if the signal value is “1”, and to “111” (−0.25) if the signal value is “0” in a similar manner. Theadder246 adds the outputs of thebit expanders244 and245.
Here, the reason why theequalizer241 provides the characteristic by the transfer function of thecoefficient3, will be described. In a case of the feedback method, the canceling sound signal output from the noise cancelingdigital filter230 includes not only the component corresponding to the external sound but also the component obtained by collecting the sound of the digital audio source output from the driver261. In other words, a characteristic according to the transfer function represented by 1/(1+β) is provided with the sound component of the digital audio source. Therefore, the characteristic of a 1+β, transfer function, which is 1/(1+β), is provided in advance with the signal of the digital audio source. Theequalizer241 provides the characteristic by the transfer function of β, from the functions. The addition of the signal by theadder246 serves equivalent to the provision of the characteristic by the transfer function of 1+β, to the digital audio source. The signal after the addition by theadder246 can take three 3-bit values, which are “010” (0.5), “000” (0), and “110” (−0.5).
Theadder247 adds the output of thebit expander234 and the output of theadder246. The signal after the addition by theadder247 can take four 3-bit values, which are “011” (0.75), “001” (0.25), “111” (−0.25) and “101” (−0.75).
The noise cancelingdigital filter230, thedelta sigma modulators232 and243, the bit expanders234,244, and245, theequalizer241, thedelayer242 and theadders246 and247 can be provided in, for example, a DSP. This DSP may be provided, for example, as a single chip component.
ThePWM conversion unit248 performs PWM modulation on the [64 Fs, 3 bits] digital signal output from theadder247. Theanalog LPF249 inputs the signal output from thePWM conversion unit248 and generates an analog audio signal. The analog audio signal generated by theanalog LPF249 is input to thepower amplifier250. Thepower amplifier250 amplifies the input audio signal and drives the driver261 corresponding to one of the ears in theheadphone260 by using its output.
Here, it is focused on a signal path of the digital audio source in the noise canceling system illustrated inFIG. 3. The signal of the digital audio source (signal passing through the delayer242) which is not provided with the characteristic β by the transfer function does not pass through the delta sigma modulator which may cause quantization noise. In other words, the signal of the digital audio source to which the characteristic β, by the transfer function is not provided is converted into an analog audio signal through thePWM conversion unit248 and theanalog LPF249 without passing through the delta sigma modulator.
Therefore, the noise canceling system according to the third embodiment of the present disclosure favorably delivers the sound of the digital audio source to the listener without degrading the audio characteristic of the digital audio source in a case where external noise is suppressed.
5. Fourth Embodiment (Feedback Method and Feed-Forward Method)
As a fourth embodiment, an example of a noise canceling system will be described, in which an audio characteristic of a digital audio source are not degraded in a noise canceling system combining a feedback method and a feed-forward method.
FIG. 4 is a diagram illustrating a configuration example of a noise canceling system according to the fourth embodiment of the present disclosure. In the following, a configuration example of the noise canceling system according to the fourth embodiment of the present disclosure will be described with reference toFIG. 4. Although the digital audio source in the noise canceling system illustrated inFIG. 4 is a DSD audio source, the present disclosure is not limited to such an example.
The noise canceling system illustrated inFIG. 4 is a combination of the noise canceling system, which is a combination of the feedback noise canceling system illustrated inFIG. 3 and a feed-forward noise canceling system. In other words, the noise canceling system illustrated inFIG. 4 is different from the feedback noise canceling system illustrated inFIG. 3 in that amicrophone271, anamplifier272,delta sigma modulators273 and275, a noise cancelingdigital filter274, and abit expander276 are added.
Themicrophone271 collects external sound (external noise), which is to be canceled, in a vicinity of theheadphone260. In the feed-forward noise canceling system, themicrophones271 are respectively provided outside a housing of each of L and R single-side channels of theheadphone260 in actual. InFIG. 4, themicrophone271 provided corresponding to one of the L channel and R channel is illustrated.
Theamplifier272 amplifies the external sound collected by themicrophone271 and an analog audio signal is obtained. The delta sigma modulator273 converts the analog audio signal output from theamplifier272 into a digital signal having same [64 Fs, 1 bit] as the digital audio source.
The noise cancelingdigital filter274 receives the digital audio signal output from thedelta sigma modulator273, that is, the digital audio signal obtained by collecting the external sound collected by themicrophone271. Then, the noise cancelingdigital filter274 uses the input digital audio signal and generates a sound signal for cancellation of sound, which is effective to cancel external sound that can be heard by reaching an ear of the wearer of theheadphone260 corresponding to the driver261 as a sound to be output from the driver261. The noise cancelingdigital filter274 is configured, for example, as an FIR filter. In the present embodiment, the noise cancelingdigital filter274 is configured as a filter whose input is [64 Fs, 1 bit] and whose output is [64 Fs, 16 bits]. Therefore, the output of the noise cancelingdigital filter274 is converted into a multi-bit.
The delta sigma modulator275 converts the quantization bit number in the [64 Fs, 16 bits] digital signal output from the noise cancelingdigital filter274 into one bit. That is, thedelta sigma modulator275 generates a [64 Fs, 1 bit] digital signal from the [64 Fs, 16 bits] digital signal output from the noise cancelingdigital filter274.
The bit expander276 converts the [64 Fs, 1 bit] digital signal output from the delta sigma modulator275 into a [64 Fs, 4 bits] digital signal in this example. In the present embodiment, the bit expanders234,244, and245 also convert a [64 Fs, 1 bit] digital signal into a [64 Fs, 4 bits] digital signal in a similar manner. In other words, each bit expander extends the digital signal having a quantization bit number of one bit into four bits so as to correspond to the addition of four digital signals.
The digital signal output from thebit expander276 is added together with the output of thebit expander234 and the output of theadder246 in theadder247.
The noise canceling system illustrated inFIG. 4 can further enhance the external noise suppression effect by combining the feed-forward noise canceling system and the feedback noise canceling system. Then, the noise canceling system according to the fourth embodiment of the present disclosure favorably delivers the sound of the digital audio source to the listener without degrading the audio characteristic of the digital audio source in a case where external noise is suppressed.
6. Fifth Embodiment (Front-Rear Insertion Feedback Method)
As a feedback noise canceling system, a method that suppresses the degradation of the quality of the audio signal while suppressing external noise by adding an audio component before and after the block providing a predetermined transfer function for noise cancellation (known as a front-rear insertion feedback method). For example, Patent Document 2 (Japanese Patent Application Laid-Open No. 2009-33309) describes a front-rear insertion feedback method.
In a fifth embodiment of the present disclosure, a noise canceling system will be described which does not degrade an audio characteristic of a digital audio source in a case where external noise is suppressed by the front-rear insertion feedback method.
FIG. 5 is a diagram illustrating a configuration example of the noise canceling system according to the fifth embodiment of the present disclosure. In the following, a configuration example of the noise canceling system according to the fifth embodiment of the present disclosure will be described with reference toFIG. 5. Although the digital audio source in the noise canceling system illustrated inFIG. 5 is a DSD audio source, the present disclosure is not limited to this example.
As illustrated inFIG. 5, the noise canceling system according to the fifth embodiment of the present disclosure includes amicrophone311, anamplifier312,delta sigma modulators313,324,325, and332,bit expanders314,326,327,328, and333,equalizers321 and322, adelayer323,adders329,330,334, and335, a noise cancelingdigital filter331, aPWM conversion unit336, ananalog LPF337, apower amplifier338, and aheadphone350.
Themicrophone311 collects sound output from thedriver351 and external sound intruding into an inside of a housing of theheadphone350, which can be cancellation targets. In the feedback noise canceling system, themicrophones311 are actually provided inside the corresponding housing for each of L and R one-side channels of theheadphone350. InFIG. 5, themicrophone311 provided corresponding to one of the L channel and R channel is illustrated.
Theamplifier312 amplifies the external sound collected by themicrophone311 and an analog audio signal is obtained.
The delta sigma modulator313 converts the analog audio signal output from theamplifier312 into a digital audio signal. The delta sigma modulator313 converts the analog audio signal output from theamplifier312 into a digital signal having same [64 Fs, 1 bit] as the digital audio source.
The bit expander314 converts the [64 Fs, 1 bit] digital signal output from the delta sigma modulator313 into a [64 Fs, 3 bits] digital signal in this example. More specifically, thebit expander314 converts the signal value into “001” (0.25) if the value of the signal is “1” and to “111” (−0.25) if the value is “0”.
Theequalizer321 is a processing block that provides the audio source with a predetermined target characteristic on a front insertion side. Also, theequalizer322 is a processing block that provides the audio source with a predetermined target characteristic on a rear insertion side. Theequalizers321 and322 convert a [64 Fs, 1 bit] digital signal into a [64 Fs, 16 bits] digital signal in this example. Thedelayer323 performs predetermined delay processing on the signal from the digital audio source in accordance with the delay due to the signal processing in theequalizers321 and322 and thedelta sigma modulators324 and325.
Here, an equalizer target characteristic EQ1 on the front insertion side and an equalizer target characteristic EQ2 on the rear insertion side both become approximately a Mid Presence Filter (hereinafter, referred to as MPF), and the characteristics are generally equalizing-adjusted. Since MPF can develop a transfer function like “1+EQ”, [64 Fs, 1 bit] of the digital audio source DSD format is branched to “1” side processing and “EQ” side processing of the target characteristic “1+EQ” and then the results are synthesized. A path passing through thedelayer323 corresponds to the former processing on the “1” side.
The delta sigma modulator324 converts the audio signal output from theequalizer321 into a digital signal having the same [64 Fs, 1 bit] as the digital audio source. The delta sigma modulator325 converts the audio signal output from theequalizer321 into a digital signal having the same [64 Fs, 1 bit] as the digital audio source.
The bit expanders326,327, and328 respectively convert the [64 Fs, 1 bit] digital signals output from thedelta sigma modulator324, thedelayer323, and the delta sigma modulator325 into [64 Fs, 3 bits] digital signals in this example. More specifically, the bit expanders326,327, and328 convert the signal value to “001” (0.25) if the signal value is “1”, and to “111” (−0.25) if the signal value is “0”, respectively.
The adder329 adds the outputs of thebit expanders326 and328. The outputs of thebit expanders326 and328 are added to achieve the target characteristic “1+EQ” described above. The signal after the addition of the adder329 can take three 3-bit values of “010” (0.5), “000” (0), and “110” (−0.5). Then, theadder330 adds the output of thebit expander314 and the output of the adder329. The signal after addition of theadder330 can take four 3-bit values of “011” (0.75), “001” (0.25), “111” (−0.25), and “101” (−0.75).
The noise cancelingdigital filter331 inputs a signal including a signal output from theadder330, that is, a digital audio signal obtained by collecting the internal sound of the housing on thedriver351 side of theheadphone350 collected by themicrophone311. Then, using the input digital audio signal, the noise cancelingdigital filter331 generates an audio signal (sound signal for cancellation) of sound, which is effective to cancel external sound that can be heard by reaching an ear of a wearer of theheadphone350 corresponding to thedriver351 as sound to be output from thedriver351. More specifically, the noise cancelingdigital filter331 performs processing for providing a predetermined transfer function −β for noise cancellation to the sound collected by themicrophone311. In the present embodiment, β is variable.
According to the present embodiment, the noise cancelingdigital filter331 is configured as a filter whose input is [64 Fs, 3 bits] and whose output is [64 Fs, 48 bits]. Therefore, the output of the noise cancelingdigital filter331 is converted into a multi-bit.
The delta sigma modulator332 converts the quantization bit number in the [64 Fs, 48 bits] digital signal output from the noise cancelingdigital filter331 into one bit. In other words, thedelta sigma modulator332 generates a [64 Fs, 1 bit] digital signal from the [64 Fs, 48 bits] digital signal output from the noise cancelingdigital filter331.
The bit expander333 converts the [64 Fs, 1 bit] digital signal output from the delta sigma modulator332 into a [64 Fs, 3 bits] digital signal in this example. More specifically, thebit expander314 converts the signal value into “001” (0.25) if the value of the signal is “1” and to “111” (−0.25) if the value is “0”.
Theadder334 adds the outputs of thebit expanders327 and328. The outputs of thebit expanders327 and328 are added to achieve the target characteristic “1+EQ” described above. The signal after the addition of the adder329 can take three 3-bit values of “010” (0.5), “000” (0), and “110” (−0.5). Then, the adder335 adds the output of thebit expander333 and the output of theadder334. The signal after addition of the adder335 can take four 3-bit values of “011” (0.75), “001” (0.25), “111” (−0.25), and “101” (−0.75).
Thedelta sigma modulators313,324,325, and332, the bit expanders314,326,327,328, and333, theequalizers321 and322, thedelayer323, theadders329,330,334, and335, and the noise cancelingdigital filter331 may be provided, for example, in the DSP. This DSP may be provided, for example, as a single chip component.
ThePWM conversion unit336 performs PWM modulation on the [64 Fs, 3 bits] digital signal output from the adder335. Then, theanalog LPF337 receives the signal output from thePWM conversion unit336 and generates an analog audio signal. The analog audio signal generated by theanalog LPF337 is input to thepower amplifier338. Thepower amplifier338 amplifies the input audio signal and drives thedriver351 corresponding to one ear in theheadphone350 by using the output.
Here, it is focused on a signal path of the digital audio source in the noise canceling system illustrated inFIG. 5. The signal of the digital audio source which is not provided with the characteristic β by the transfer function (that is, the signal via thedelayer323 and theadders334 and335) does not pass through the delta sigma modulator which may cause quantization noise. In other words, the signal of the digital audio source, which is not provided with the characteristic β by the transfer function, is converted to an analog audio signal through thePWM conversion unit336 and theanalog LPF337 without being passed through the delta sigma modulator.
Therefore, the noise canceling system according to the fifth embodiment of the present disclosure favorably delivers the sound of the digital audio source to the listener without degrading the audio characteristic of the digital audio source in a case where external noise is suppressed.
7. Summary
As described above, according to the embodiments of the present disclosure, a noise canceling system is provided which can favorably deliver sound of a digital audio source to a listener without degrading an audio characteristic of the digital audio source in a case where external noise is suppressed.
It is possible to create a computer program that causes hardware included in each device, such as a CPU, a ROM, and a RAM, to provide the same function as the above described configuration of each device. Also, a storage medium storing such computer program can be provided. In addition, by configuring each functional block illustrated in the functional block diagram by hardware, the series of processes can be realized by the hardware.
Although the preferred embodiments of the present disclosure have been described in detail with reference to the accompanying drawings, the technical scope of the present disclosure is not limited to such examples. It is obvious that persons having ordinary knowledge in the technical field of the present disclosure can conceive various changes or modifications within the scope of the technical idea described in the claims and it is naturally understood to those changes and modifications belong to the technical scope of the present disclosure.
In addition, the effects described in this specification are merely illustrative or exemplary and does not set any limitation. In other words, the technique according to the present disclosure can provide other effects obvious to those skilled in the art from the description of the present specification together with the above described effects or in addition to the above effects.
Note that the following configurations are also within the technical scope of the present disclosure.
(1)
A signal processing apparatus including:
an A/D converter configured to output a digital signal having a predetermined sampling frequency and quantization bit number a, the A/D converter including a first delta sigma modulator that performs a first delta sigma modulation process on an input analog signal;
a filter unit configured to pass an output of the A/D converter through a digital filter provided with a predetermined filter characteristic and output a digital signal having the sampling frequency and a quantization bit number b;
a second delta sigma modulator configured to perform a second delta sigma modulation process on an output of the filter unit and output a digital signal having the sampling frequency and the quantization bit number a; and
an addition unit configured to add an output of the second delta sigma modulator and an input digital signal having the sampling frequency and the quantization bit number a.
(2)
The signal processing apparatus according to (1), in which the analog signal is sound collected by a microphone provided at a predetermined position in a headphone.
(3)
The signal processing apparatus according to (2), in which the predetermined filter characteristic is a filter characteristic for performing a feed-forward noise reduction process for the headphone.
(4)
The signal processing apparatus according to any one of (1) to (3), in which the input digital signal is a DSD audio signal.
(5)
A signal processing apparatus including:
an A/D converter configured to output a digital signal having a predetermined sampling frequency and a quantization bit number a, the A/D converter including a first delta sigma modulator that performs a first delta sigma modulation process on an input analog signal;
a filter unit configured to pass an output of the A/D converter through a digital filter provided with a predetermined filter characteristic and output a digital signal having the sampling frequency and a quantization bit number b;
a second delta sigma modulator configured to perform a second delta sigma modulation process on an output of the filter unit and output a digital signal having the sampling frequency and the quantization bit number a;
a first bit expander configured to expand the quantization bit number from a to c for an output of the second delta sigma modulator; and
a first addition unit configured to add an output of the first bit expander and an input digital signal having the sampling frequency and a quantization bit number c.
(6)
The signal processing apparatus according to (5), in which the analog signal is sound collected by a microphone provided at a predetermined position in a headphone.
(7)
The signal processing apparatus according to (6), in which the predetermined filter characteristic is a filter characteristic for performing a feedback noise reduction process for the headphone.
(8)
The signal processing apparatus according to (6) or (7), in which the digital signal, the digital signal having passed through the digital filter provided with the filter characteristic for performing the feed-forward noise reduction process for the headphone, is also added to the first addition unit.
(9)
The signal processing apparatus according to any one of (5) to (8), further including:
an equalizer unit configured to equalize the input digital signal with a predetermined target characteristic;
a third delta sigma modulator configured to perform a third delta sigma modulation process on an output of the equalizer unit and output a digital signal having the sampling frequency and the quantization bit number a;
a delay unit configured to provide the input digital signal with a delay equivalent to processing delay in the equalizer unit and the third delta sigma modulator;
a second bit expander configured to expand the quantization bit number from a to c for an output of the third delta sigma modulator;
a third bit expander configured to expand the quantization bit number from a to c for an output of the delay unit; and
a second adder configured to add outputs of the second bit expander and the third bit expander and output a result to the first addition unit.
(10)
The signal processing apparatus according to any one of (5) to (9), in which the input digital signal is a DSD audio signal.
(11)
A signal processing apparatus including:
a first delta sigma modulation unit configured to perform a first delta sigma modulation process on an input analog signal, generate a digital signal having a predetermined sampling frequency and a quantization bit number a, and output the digital signal after expanding its quantization bit number from a to c;
a first equalizer unit configured to generate a first equalized signal by equalizing an input digital signal having the sampling frequency and a quantization bit number c with a first target characteristic, perform a delta sigma modulation process on the first equalized signal, generate a digital signal having the predetermined sampling frequency and the quantization bit number a, and output the digital signal after expanding its quantization bit number from a to c;
a second equalizer unit configured to generate a second equalized signal by equalizing an input digital signal having the sampling frequency and the quantization bit number c with a second target characteristic, perform the delta sigma modulation process on the second equalized signal, generate a digital signal having the predetermined sampling frequency and the quantization bit number a, and output the digital signal after expanding its quantization bit number from a to c;
a delay unit configured to provide the input digital signal with a delay equivalent to processing delay in the first equalizer unit or the second equalizer unit, and output the signal after expanding its quantization bit number from a to c;
a first addition unit configured to add outputs of the first delta sigma modulation unit, the delay unit, and the first equalizer unit;
a filter unit configured to pass an output of the first addition unit through a digital filter provided with a predetermined filter characteristic and output a digital signal having the sampling frequency and a quantization bit number b;
a second delta sigma modulation unit configured to perform a delta sigma modulation process on an output of the filter unit, generate a digital signal having the sampling frequency and the quantization bit number a, and output the digital signal after expanding its quantization bit number from a to c; and a second addition unit configured to add outputs of the second delta sigma modulation unit, the delay unit, and the second equalizer unit.
(12)
The signal processing apparatus according to (11), in which the analog signal is sound collected by a microphone provided at a predetermined position in a headphone.
(13)
The signal processing apparatus according to (12), in which the predetermined filter characteristic is a filter characteristic for performing a feedback noise reduction process for the headphone.
(14)
The signal processing apparatus according to any one of (11) to (13), in which the input digital signal is a DSD audio signal.
(15)
A signal processing method including:
outputting a digital signal having a predetermined sampling frequency and a quantization bit number a by performing a first delta sigma modulation process on an input analog signal;
passing a digital signal having the predetermined sampling frequency and the quantization bit number a through a digital filter provided with a predetermined filter characteristic and outputting a digital signal having the sampling frequency and a quantization bit number b;
outputting a digital signal having the sampling frequency and the quantization bit number a by performing a second delta sigma modulation process on the digital signal having the quantization bit number b; and
adding an output of the second delta sigma modulation process and an input digital signal having the sampling frequency and the quantization bit number a.
(16)
A signal processing method including:
outputting a digital signal having a predetermined sampling frequency and a quantization bit number a by performing a first delta sigma modulation process on an input analog signal;
passing a digital signal having the predetermined sampling frequency and the quantization bit number a through a digital filter provided with a predetermined filter characteristic and outputting a digital signal having the sampling frequency and a quantization bit number b;
outputting a digital signal having the sampling frequency and the quantization bit number a by performing a second delta sigma modulation process on an output of the digital signal having the sampling frequency and the quantization bit number b;
expanding the quantization bit number from a to c for an output of the second delta sigma modulation process; and
adding the digital signal having an expanded quantization bit number c and an input digital signal having the sampling frequency and a quantization bit number c.
(17)
A signal processing method including:
performing a first delta sigma modulation process where a digital signal having a predetermined sampling frequency and a quantization bit number a is generated by performing a delta sigma modulation process on an input analog signal, and the digital signal is output after expanding its quantization bit number from a to c;
performing a first equalization process where a first equalized signal is generated by equalizing an input digital signal having the sampling frequency and a quantization bit number c with a first target characteristic, a delta sigma modulation process is performed on the first equalized signal, a digital signal having the predetermined sampling frequency and the quantization bit number a is generated, and the digital signal is output after expanding its quantization bit number from a to c;
performing a second equalization process where a second equalized signal is generated by equalizing an input digital signal having the sampling frequency and a quantization bit number c with a second target characteristic, a delta sigma modulation process is performed on the second equalized signal, a digital signal having the predetermined sampling frequency and the quantization bit number a is generated, and the digital signal is output after expanding its quantization bit number from a to c;
performing a delay process where the input digital signal is provided with a delay equivalent to processing delay in the first equalization process or the second equalization process, and the signal is output after expanding its quantization bit number from a to c;
performing a first addition process where outputs of the first delta sigma modulation process, the delay process, and the first equalization process;
performing a filter process where an output of the first addition process is passed through a digital filter provided with a predetermined filter characteristic and, a digital signal having the sampling frequency and a quantization bit number b is output;
performing a second delta sigma modulation process where a digital signal having the sampling frequency and the quantization bit number a is generated by performing a delta sigma modulation process on an output of the filter process and the digital signal is output after expanding its quantization bit number from a to c; and performing a first addition process where outputs of the second delta sigma modulation process, the delay process, and the second equalization process.
(18)
A computer program that causes a computer to execute:
outputting a digital signal having a predetermined sampling frequency and a quantization bit number a by performing a first delta sigma modulation process on an input analog signal;
passing a digital signal having the predetermined sampling frequency and the quantization bit number a through a digital filter provided with a predetermined filter characteristic and outputting a digital signal having the sampling frequency and a quantization bit number b;
outputting a digital signal having the sampling frequency and the quantization bit number a by performing a second delta sigma modulation process on the digital signal having the quantization bit number b; and
adding an output of the second delta sigma modulation process and an input digital signal having the sampling frequency and the quantization bit number a.
(19)
A computer program that causes a computer to execute:
outputting a digital signal having a predetermined sampling frequency and a quantization bit number a by performing a first delta sigma modulation process on an input analog signal;
passing a digital signal having the predetermined sampling frequency and the quantization bit number a through a digital filter provided with a predetermined filter characteristic and outputting a digital signal having the sampling frequency and a quantization bit number b;
outputting a digital signal having the sampling frequency and the quantization bit number a by performing a second delta sigma modulation process on an output of the digital signal having the sampling frequency and the quantization bit number b;
expanding the quantization bit number from a to c for an output of the second delta sigma modulation process; and
adding the digital signal having an expanded quantization bit number c and an input digital signal having the sampling frequency and a quantization bit number c.
(20)
A computer program that causes a computer to execute:
performing a first delta sigma modulation process where a digital signal having a predetermined sampling frequency and a quantization bit number a is generated by performing a delta sigma modulation process on an input analog signal, and the digital signal is output after expanding its quantization bit number from a to c;
performing a first equalization process where a first equalized signal is generated by equalizing an input digital signal having the sampling frequency and a quantization bit number c with a first target characteristic, a delta sigma modulation process is performed on the first equalized signal, a digital signal having the predetermined sampling frequency and the quantization bit number a is generated, and the digital signal is output after expanding its quantization bit number from a to c;
performing a second equalization process where a second equalized signal is generated by equalizing an input digital signal having the sampling frequency and a quantization bit number c with a second target characteristic, a delta sigma modulation process is performed on the second equalized signal, a digital signal having the predetermined sampling frequency and the quantization bit number a is generated, and the digital signal is output after expanding its quantization bit number from a to c;
performing a delay process where the input digital signal is provided with a delay equivalent to processing delay in the first equalization process or the second equalization process, and the signal is output after expanding its quantization bit number from a to c;
performing a first addition process where outputs of the first delta sigma modulation process, the delay process, and the first equalization process;
performing a filter process where an output of the first addition process is passed through a digital filter provided with a predetermined filter characteristic, and a digital signal having the sampling frequency and a quantization bit number b is output;
performing a second delta sigma modulation process where a digital signal having the sampling frequency and the quantization bit number a is generated by performing a delta sigma modulation process on an output of the filter process and the digital signal is output after expanding its quantization bit number from a to c; and
performing a first addition process where outputs of the second delta sigma modulation process, the delay process, and the second equalization process.
REFERENCE SIGNS LIST
  • 111 Microphone
  • 112 Amplifier
  • 134 Adder
  • 140 Power amplifier
  • 150 Headphone
  • 151 Driver
  • 152 Driver

Claims (20)

The invention claimed is:
1. A signal processing apparatus comprising:
an A/D converter configured to output a first digital signal having a predetermined sampling frequency and quantization bit number a, the A/D converter including a first delta sigma modulator that performs a first delta sigma modulation process on an input analog signal;
a filter unit configured to pass the first digital signal through a digital filter provided with a predetermined filter characteristic and output a second digital signal having a sampling frequency and a quantization bit number b;
a second delta sigma modulator configured to perform a second delta sigma modulation process on the second digital signal and output a third digital signal having the predetermined sampling frequency and the quantization bit number a; and
an addition unit configured to add the third digital signal and an input digital signal having the predetermined sampling frequency and the quantization bit number a.
2. The signal processing apparatus according toclaim 1, wherein the analog signal is sound collected by a microphone provided at a predetermined position in a headphone.
3. The signal processing apparatus according toclaim 2, wherein the predetermined filter characteristic is a filter characteristic for performing a feed-forward noise reduction process for the headphone.
4. The signal processing apparatus according toclaim 1, wherein the input digital signal is a DSD (direct stream digital) audio signal.
5. A signal processing apparatus comprising:
an A/D converter configured to output a first digital signal having a predetermined sampling frequency and a quantization bit number a, the A/D converter including a first delta sigma modulator that performs a first delta sigma modulation process on an input analog signal;
a filter unit configured to pass the first digital signal through a digital filter provided with a predetermined filter characteristic and output a second digital signal having a sampling frequency and a quantization bit number b;
a second delta sigma modulator configured to perform a second delta sigma modulation process on the second digital signal and output a third digital signal having the predetermined sampling frequency and the quantization bit number a;
a first bit expander configured to expand the quantization bit number of the third digital signal from a to c to generate a fourth digital signal; and
a first addition unit configured to add the fourth digital signal and an input digital signal having the sampling frequency and a quantization bit number c.
6. The signal processing apparatus according toclaim 5, wherein the analog signal is sound collected by a microphone provided at a predetermined position in a headphone.
7. The signal processing apparatus according toclaim 6, wherein the predetermined filter characteristic is a filter characteristic for performing a feedback noise reduction process for the headphone.
8. The signal processing apparatus according toclaim 6, wherein the digital signal, the digital signal having passed through the digital filter provided with the filter characteristic for performing the feed-forward noise reduction process for the headphone, is also added to the first addition unit.
9. The signal processing apparatus according toclaim 5, further comprising:
an equalizer unit configured to equalize the input digital signal with a predetermined target characteristic;
a third delta sigma modulator configured to perform a third delta sigma modulation process on an output of the equalizer unit and output the second digital signal having the predetermined sampling frequency and the quantization bit number a;
a delay unit configured to provide the input digital signal with a delay equivalent to processing delay in the equalizer unit and the third delta sigma modulator;
a second bit expander configured to expand the quantization bit number from a to c for an output of the third delta sigma modulator;
a third bit expander configured to expand the quantization bit number from a to c for an output of the delay unit; and
a second adder configured to add outputs of the second bit expander and the third bit expander and output a result to the first addition unit.
10. The signal processing apparatus according toclaim 5, wherein the input digital signal is a DSD (direct stream digital) audio signal.
11. A signal processing apparatus comprising:
a first delta sigma modulation unit configured to perform a first delta sigma modulation process on an input analog signal to:
generate a first digital signal having a predetermined sampling frequency and a quantization bit number,
expand the quantization bit number of the first digital signal from a to c, and
output a second digital signal;
a first equalizer unit configured to perform a first equalizing process on an input digital signal having the predetermined sampling frequency and the quantization bit number a with a first target characteristic to:
generate a first equalized signal,
perform a delta sigma modulation process on the first equalized signal,
generate a third digital signal having the predetermined sampling frequency and the quantization bit number a,
expand the quantization bit number of the third digital signal from a to c, and
output a fourth digital signal;
a second equalizer unit configured to perform a second equalizing process on the input digital signal with a second target characteristic to:
generate a second equalized signal,
perform a delta sigma modulation process on the second equalized signal,
generate a fifth digital signal having the predetermined sampling frequency and the quantization bit number a,
expand the quantization bit number of the fifth digital signal from a to c, and
output a sixth digital signal;
a delay unit configured to provide the input digital signal with a delay equivalent to processing delay in the first equalizer unit or the second equalizer unit to:
generate a seventh digital signal;
expand the quantization bit number of the seventh digital signal from a to c, and
output an eighth digital signal;
a first addition unit configured to add the second digital signal and the fourth digital to generate a ninth digital signal;
a filter unit configured to pass the ninth digital signal through a digital filter provided with a predetermined filter characteristic and output a tenth digital signal having the predetermined sampling frequency and a quantization bit number b;
a second delta sigma modulation unit configured to perform a delta sigma modulation process on the tenth digital signal to:
generate an eleventh digital signal having the predetermined sampling frequency and the quantization bit number a,
expand the quantization bit number of the eleventh digital signal from a to c, and
output a twelfth digital signal; and
a second addition unit configured to add the sixth digital signal, the eighth digital signal, and the twelfth digital signal.
12. The signal processing apparatus according toclaim 11, wherein the analog signal is sound collected by a microphone provided at a predetermined position in a headphone.
13. The signal processing apparatus according toclaim 11, wherein the predetermined filter characteristic is a filter characteristic for performing a feedback noise reduction process for the headphone.
14. The signal processing apparatus according toclaim 11, wherein the input digital signal is a DSD (direct stream digital) audio signal.
15. A signal processing method comprising:
outputting a first digital signal having a predetermined sampling frequency and a quantization bit number a by performing a first delta sigma modulation process on an input analog signal;
passing the first digital signal having the predetermined sampling frequency and the quantization bit number a through a digital filter provided with a predetermined filter characteristic and outputting a second digital signal having the sampling frequency and a quantization bit number b;
outputting a third digital signal having the predetermined sampling frequency and the quantization bit number a by performing a second delta sigma modulation process on the second digital signal having the quantization bit number b; and
adding the third digital signal and an input digital signal having the sampling frequency and the quantization bit number a.
16. A signal processing method comprising:
outputting a first digital signal having a predetermined sampling frequency and a quantization bit number a by performing a first delta sigma modulation process on an input analog signal;
passing the first digital signal having the predetermined sampling frequency and the quantization bit number a through a digital filter provided with a predetermined filter characteristic and outputting a second digital signal having the predetermining sampling frequency and a quantization bit number b;
outputting a third digital signal having the predetermined sampling frequency and the quantization bit number a by performing a second delta sigma modulation process on an output of the digital signal having the sampling frequency and the quantization bit number b;
expanding the quantization bit number of the third digital signal from a to c for an output of the second delta sigma modulation process to generate a fourth digital signal; and
adding the fourth digital signal having an expanded quantization bit number c and an input digital signal having the sampling frequency and a quantization bit number c.
17. A non-transitory computer-readable storage medium storing a computer program that causes a computer to execute:
outputting a first digital signal having a predetermined sampling frequency and a quantization bit number a by performing a first delta sigma modulation process on an input analog signal;
passing the first digital signal having the predetermined sampling frequency and the quantization bit number a through a digital filter provided with a predetermined filter characteristic and outputting a second digital signal having the sampling frequency and a quantization bit number b;
outputting a third digital signal having the predetermined sampling frequency and the quantization bit number a by performing a second delta sigma modulation process on the second digital signal having the quantization bit number b; and
adding the third digital signal and an input digital signal having the sampling frequency and the quantization bit number a.
18. A non-transitory computer-readable storage medium storing a computer program that causes a computer to execute:
outputting a first digital signal having a predetermined sampling frequency and a quantization bit number a by performing a first delta sigma modulation process on an input analog signal;
passing the first digital signal having the predetermined sampling frequency and the quantization bit number a through a digital filter provided with a predetermined filter characteristic and outputting a second digital signal having the predetermined sampling frequency and a quantization bit number b;
outputting a third digital signal having the predetermined sampling frequency and the quantization bit number a by performing a second delta sigma modulation process on an output of the digital signal having the sampling frequency and the quantization bit number b;
expanding the quantization bit number of the third digital signal from a to c for an output of the second delta sigma modulation process to generate a fourth digital signal; and
adding the fourth digital signal having an expanded quantization bit number c and an input digital signal having the sampling frequency and a quantization bit number c.
19. A signal processing method comprising:
performing a first delta sigma modulation unit configured to perform a first delta sigma modulation process on an input analog signal to:
generate a first digital signal having a predetermined sampling frequency and a quantization bit number,
expand the quantization bit number of the first digital signal from a to c, and
output a second digital signal;
performing a first equalizing process on an input digital signal having the predetermined sampling frequency and the quantization bit number a with a first target characteristic to:
generate a first equalized signal,
perform a delta sigma modulation process on the first equalized signal,
generate a third digital signal having the predetermined sampling frequency and the quantization bit number a,
expand the quantization bit number of the third digital signal from a to c, and
output a fourth digital signal;
performing a second equalizing process on the input digital signal with a second target characteristic to:
generate a second equalized signal,
perform a delta sigma modulation process on the second equalized signal,
generate a fifth digital signal having the predetermined sampling frequency and the quantization bit number a,
expand the quantization bit number of the fifth digital signal from a to c, and
output a sixth digital signal;
providing the input digital signal with a delay equivalent to processing delay to:
generate a seventh digital signal;
expand the quantization bit number of the seventh digital signal from a to c, and
output an eighth digital signal;
adding the second digital signal and the fourth digital to generate a ninth digital signal;
passing the ninth digital signal through a digital filter provided with a predetermined filter characteristic and output a tenth digital signal having the predetermined sampling frequency and a quantization bit number b;
performing a delta sigma modulation process on the tenth digital signal to:
generate an eleventh digital signal having the predetermined sampling frequency and the quantization bit number a,
expand the quantization bit number of the eleventh digital signal from a to c, and
output a twelfth digital signal; and
adding the sixth digital signal, the eighth digital signal, and the twelfth digital signal.
20. A non-transitory computer-readable storage medium having computer readable instructions stored thereon that, when executed by a computer, cause the computer to perform the steps of:
performing a first delta sigma modulation unit configured to perform a first delta sigma modulation process on an input analog signal to:
generate a first digital signal having a predetermined sampling frequency and a quantization bit number,
expand the quantization bit number of the first digital signal from a to c, and
output a second digital signal;
performing a first equalizing process on an input digital signal having the predetermined sampling frequency and the quantization bit number a with a first target characteristic to:
generate a first equalized signal,
perform a delta sigma modulation process on the first equalized signal,
generate a third digital signal having the predetermined sampling frequency and the quantization bit number a,
expand the quantization bit number of the third digital signal from a to c, and
output a fourth digital signal;
performing a second equalizing process on the input digital signal with a second target characteristic to:
generate a second equalized signal,
perform a delta sigma modulation process on the second equalized signal,
generate a fifth digital signal having the predetermined sampling frequency and the quantization bit number a,
expand the quantization bit number of the fifth digital signal from a to c, and
output a sixth digital signal;
providing the input digital signal with a delay equivalent to processing delay to:
generate a seventh digital signal;
expand the quantization bit number of the seventh digital signal from a to c, and
output an eighth digital signal;
adding the second digital signal and the fourth digital to generate a ninth digital signal;
passing the ninth digital signal through a digital filter provided with a predetermined filter characteristic and output a tenth digital signal having the predetermined sampling frequency and a quantization bit number b;
performing a delta sigma modulation process on the tenth digital signal to:
generate an eleventh digital signal having the predetermined sampling frequency and the quantization bit number a,
expand the quantization bit number of the eleventh digital signal from a to c, and
output a twelfth digital signal; and
adding the sixth digital signal, the eighth digital signal, and the twelfth digital signal.
US16/480,3812017-01-312017-12-11Signal processing apparatus, signal processing method, and computer programActiveUS10896668B2 (en)

Applications Claiming Priority (3)

Application NumberPriority DateFiling DateTitle
JP2017-0158072017-01-31
JP20170158072017-01-31
PCT/JP2017/044374WO2018142770A1 (en)2017-01-312017-12-11Signal processing device, signal processing method, and computer program

Publications (2)

Publication NumberPublication Date
US20190385585A1 US20190385585A1 (en)2019-12-19
US10896668B2true US10896668B2 (en)2021-01-19

Family

ID=63039519

Family Applications (1)

Application NumberTitlePriority DateFiling Date
US16/480,381ActiveUS10896668B2 (en)2017-01-312017-12-11Signal processing apparatus, signal processing method, and computer program

Country Status (6)

CountryLink
US (1)US10896668B2 (en)
EP (1)EP3579225A4 (en)
JP (1)JP7020432B2 (en)
KR (1)KR20190113778A (en)
CN (1)CN110226200A (en)
WO (1)WO2018142770A1 (en)

Families Citing this family (3)

* Cited by examiner, † Cited by third party
Publication numberPriority datePublication dateAssigneeTitle
US10175931B2 (en)2012-11-022019-01-08Sony CorporationSignal processing device and signal processing method
US11303295B1 (en)*2020-11-152022-04-12xMEMS Labs, Inc.SDM encoder and related signal processing system
CN117526957B (en)*2024-01-042024-03-19秦玄汉(苏州)信息科技有限公司Analog-to-digital converter with optimal quantization bit number

Citations (41)

* Cited by examiner, † Cited by third party
Publication numberPriority datePublication dateAssigneeTitle
EP0335468A1 (en)1988-03-241989-10-04Birch Wood Acoustics Nederland B.V.Electro-acoustical system
EP0735796A2 (en)1995-03-301996-10-02Kabushiki Kaisha TimewareMethod and apparatus for reproducing three-dimensional virtual space sound
US5592559A (en)1991-08-021997-01-07Sharp Kabushiki KaishaSpeaker driving circuit
EP0989543A2 (en)1998-09-252000-03-29Sony CorporationSound effect adding apparatus
US6608903B1 (en)1999-08-172003-08-19Yamaha CorporationSound field reproducing method and apparatus for the same
JP2003323179A (en)2002-02-272003-11-14Yamaha CorpMethod and instrument for measuring impulse response, and method and device for reproducing sound field
JP2006085214A (en)2004-09-142006-03-30Noritsu Koki Co Ltd Photo processing device
US20060109988A1 (en)2004-10-282006-05-25Metcalf Randall BSystem and method for generating sound events
US20070025560A1 (en)2005-08-012007-02-01Sony CorporationAudio processing method and sound field reproducing system
JP2007124023A (en)2005-10-252007-05-17Sony CorpMethod of reproducing sound field, and method and device for processing sound signal
US7233673B1 (en)1998-04-232007-06-19Industrial Research LimitedIn-line early reflection enhancement system for enhancing acoustics
US20080056517A1 (en)2002-10-182008-03-06The Regents Of The University Of CaliforniaDynamic binaural sound capture and reproduction in focued or frontal applications
US20080186218A1 (en)2007-02-052008-08-07Sony CorporationSignal processing apparatus and signal processing method
JP2008227773A (en)2007-03-092008-09-25Advanced Telecommunication Research Institute International Acoustic space sharing device
JP2008250270A (en)2007-03-022008-10-16Sony CorpSignal processing apparatus and signal processing method
US20090010443A1 (en)2007-07-062009-01-08Sda Software Design Ahnert GmbhMethod and Device for Determining a Room Acoustic Impulse Response in the Time Domain
JP2009033309A (en)2007-07-252009-02-12Sony CorpSignal processor, signal processing method, program, and noise canceling system
US20100027805A1 (en)2008-07-302010-02-04Fujitsu LimitedTransfer function estimating device, noise suppressing apparatus and transfer function estimating method
US20100150359A1 (en)2008-06-302010-06-17Constellation Productions, Inc.Methods and Systems for Improved Acoustic Environment Characterization
EP2239728A2 (en)2009-04-092010-10-13Harman International Industries, IncorporatedSystem for active noise control based on audio system output
JP4725234B2 (en)2005-08-052011-07-13ソニー株式会社 Sound field reproduction method, sound signal processing method, sound signal processing apparatus
JP2011138151A (en)2011-02-152011-07-14Sony CorpVoice signal processing method and sound field reproduction system
JP4735108B2 (en)2005-08-012011-07-27ソニー株式会社 Audio signal processing method, sound field reproduction system
JP4775487B2 (en)2009-11-242011-09-21ソニー株式会社 Audio signal processing method and audio signal processing apparatus
US8094046B2 (en)*2007-03-022012-01-10Sony CorporationSignal processing apparatus and signal processing method
JP4883197B2 (en)2010-02-152012-02-22ソニー株式会社 Audio signal processing method, sound field reproduction system
US20120093320A1 (en)2010-10-132012-04-19Microsoft CorporationSystem and method for high-precision 3-dimensional audio for augmented reality
US8165312B2 (en)*2006-04-122012-04-24Wolfson Microelectronics PlcDigital circuit arrangements for ambient noise-reduction
US20120155666A1 (en)*2010-12-162012-06-21Nair Vijayakumaran VAdaptive noise cancellation
US20120155667A1 (en)*2010-12-162012-06-21Nair Vijayakumaran VAdaptive noise cancellation
US20120307048A1 (en)2011-05-302012-12-06Sony Ericsson Mobile Communications AbSensor-based placement of sound in video recording
US20130272548A1 (en)2012-04-132013-10-17Qualcomm IncorporatedObject recognition using multi-modal matching scheme
US20140098964A1 (en)2012-10-042014-04-10Siemens CorporationMethod and Apparatus for Acoustic Area Monitoring by Exploiting Ultra Large Scale Arrays of Microphones
US20140126758A1 (en)2011-06-242014-05-08Bright Minds Holding B.V.Method and device for processing sound data
US8848935B1 (en)*2009-12-142014-09-30Audience, Inc.Low latency active noise cancellation system
US8953813B2 (en)*2010-12-012015-02-10Dialog Semiconductor GmbhReduced delay digital active noise cancellation
US20150043756A1 (en)2011-10-142015-02-12Juha Petteri OjanperaAudio scene mapping apparatus
US20150124167A1 (en)2012-04-052015-05-07Juha Henrik ArrasvuoriFlexible spatial audio capture apparatus
EP2879402A1 (en)2012-07-272015-06-03Sony CorporationInformation processing system and storage medium
US20150286463A1 (en)2012-11-022015-10-08Sony CorporationSignal processing device and signal processing method
US20150296290A1 (en)2012-11-022015-10-15Sony CorporationSignal processing device, signal processing method, measurement method, and measurement device

Patent Citations (50)

* Cited by examiner, † Cited by third party
Publication numberPriority datePublication dateAssigneeTitle
EP0335468A1 (en)1988-03-241989-10-04Birch Wood Acoustics Nederland B.V.Electro-acoustical system
US5592559A (en)1991-08-021997-01-07Sharp Kabushiki KaishaSpeaker driving circuit
EP0735796A2 (en)1995-03-301996-10-02Kabushiki Kaisha TimewareMethod and apparatus for reproducing three-dimensional virtual space sound
US7233673B1 (en)1998-04-232007-06-19Industrial Research LimitedIn-line early reflection enhancement system for enhancing acoustics
EP0989543A2 (en)1998-09-252000-03-29Sony CorporationSound effect adding apparatus
JP2000099061A (en)1998-09-252000-04-07Sony CorpEffect sound adding device
US6608903B1 (en)1999-08-172003-08-19Yamaha CorporationSound field reproducing method and apparatus for the same
JP2003323179A (en)2002-02-272003-11-14Yamaha CorpMethod and instrument for measuring impulse response, and method and device for reproducing sound field
US20080056517A1 (en)2002-10-182008-03-06The Regents Of The University Of CaliforniaDynamic binaural sound capture and reproduction in focued or frontal applications
JP2006085214A (en)2004-09-142006-03-30Noritsu Koki Co Ltd Photo processing device
US20060109988A1 (en)2004-10-282006-05-25Metcalf Randall BSystem and method for generating sound events
JP4735108B2 (en)2005-08-012011-07-27ソニー株式会社 Audio signal processing method, sound field reproduction system
US20070025560A1 (en)2005-08-012007-02-01Sony CorporationAudio processing method and sound field reproducing system
JP4674505B2 (en)2005-08-012011-04-20ソニー株式会社 Audio signal processing method, sound field reproduction system
JP4725234B2 (en)2005-08-052011-07-13ソニー株式会社 Sound field reproduction method, sound signal processing method, sound signal processing apparatus
JP2007124023A (en)2005-10-252007-05-17Sony CorpMethod of reproducing sound field, and method and device for processing sound signal
US8165312B2 (en)*2006-04-122012-04-24Wolfson Microelectronics PlcDigital circuit arrangements for ambient noise-reduction
US20080186218A1 (en)2007-02-052008-08-07Sony CorporationSignal processing apparatus and signal processing method
JP2008193421A (en)2007-02-052008-08-21Sony CorpImage processor and image processing method
EP1970901A2 (en)2007-02-052008-09-17Sony CorporationSignal processing apparatus and signal processing method
JP2008250270A (en)2007-03-022008-10-16Sony CorpSignal processing apparatus and signal processing method
US8094046B2 (en)*2007-03-022012-01-10Sony CorporationSignal processing apparatus and signal processing method
JP2008227773A (en)2007-03-092008-09-25Advanced Telecommunication Research Institute International Acoustic space sharing device
US20090010443A1 (en)2007-07-062009-01-08Sda Software Design Ahnert GmbhMethod and Device for Determining a Room Acoustic Impulse Response in the Time Domain
JP2009033309A (en)2007-07-252009-02-12Sony CorpSignal processor, signal processing method, program, and noise canceling system
US20100150359A1 (en)2008-06-302010-06-17Constellation Productions, Inc.Methods and Systems for Improved Acoustic Environment Characterization
US20100027805A1 (en)2008-07-302010-02-04Fujitsu LimitedTransfer function estimating device, noise suppressing apparatus and transfer function estimating method
US20100260345A1 (en)2009-04-092010-10-14Harman International Industries, IncorporatedSystem for active noise control based on audio system output
JP2010244045A (en)2009-04-092010-10-28Harman Internatl Industries Inc Active noise control system based on audio system output
EP2239728A2 (en)2009-04-092010-10-13Harman International Industries, IncorporatedSystem for active noise control based on audio system output
JP4775487B2 (en)2009-11-242011-09-21ソニー株式会社 Audio signal processing method and audio signal processing apparatus
US8848935B1 (en)*2009-12-142014-09-30Audience, Inc.Low latency active noise cancellation system
JP4883197B2 (en)2010-02-152012-02-22ソニー株式会社 Audio signal processing method, sound field reproduction system
US20120093320A1 (en)2010-10-132012-04-19Microsoft CorporationSystem and method for high-precision 3-dimensional audio for augmented reality
US8953813B2 (en)*2010-12-012015-02-10Dialog Semiconductor GmbhReduced delay digital active noise cancellation
US20120155666A1 (en)*2010-12-162012-06-21Nair Vijayakumaran VAdaptive noise cancellation
US20120155667A1 (en)*2010-12-162012-06-21Nair Vijayakumaran VAdaptive noise cancellation
JP2011138151A (en)2011-02-152011-07-14Sony CorpVoice signal processing method and sound field reproduction system
US20120307048A1 (en)2011-05-302012-12-06Sony Ericsson Mobile Communications AbSensor-based placement of sound in video recording
US20140126758A1 (en)2011-06-242014-05-08Bright Minds Holding B.V.Method and device for processing sound data
US20150043756A1 (en)2011-10-142015-02-12Juha Petteri OjanperaAudio scene mapping apparatus
US20150124167A1 (en)2012-04-052015-05-07Juha Henrik ArrasvuoriFlexible spatial audio capture apparatus
US20130272548A1 (en)2012-04-132013-10-17Qualcomm IncorporatedObject recognition using multi-modal matching scheme
EP2879402A1 (en)2012-07-272015-06-03Sony CorporationInformation processing system and storage medium
US20140098964A1 (en)2012-10-042014-04-10Siemens CorporationMethod and Apparatus for Acoustic Area Monitoring by Exploiting Ultra Large Scale Arrays of Microphones
US20150286463A1 (en)2012-11-022015-10-08Sony CorporationSignal processing device and signal processing method
US20150296290A1 (en)2012-11-022015-10-15Sony CorporationSignal processing device, signal processing method, measurement method, and measurement device
US9602916B2 (en)2012-11-022017-03-21Sony CorporationSignal processing device, signal processing method, measurement method, and measurement device
US10175931B2 (en)2012-11-022019-01-08Sony CorporationSignal processing device and signal processing method
US20190114136A1 (en)2012-11-022019-04-18Sony CorporationSignal processing device and signal processing method

Non-Patent Citations (17)

* Cited by examiner, † Cited by third party
Title
European Communication pursuant to Article 94(3) EPC dated Jan. 30, 2019 in connection with European Application No. 13852010.1.
European Summons to attend oral proceedings pursuant to Rule 115(1) EPC dated Oct. 17, 2019 in connection with European Application No. 13852010.1.
Extended European Search Report dated Jan. 17, 2020 in connection with European Application No. 17894910.3.
Extended European Search Report dated Jun. 3, 2016 in connection with European Application No. 13852010.1.
Extended European Search Report dated Sep. 16, 2016 in connection with European Application No. 13850571.4.
International Preliminary Report on Patentability and English translation thereof dated Aug. 15, 2019 in connection with International Application No. PCT/JP2017/044374.
International Preliminary Report on Patentability and English translation thereof dated May 14, 2015 in connection with Application No. PCT/JP2013/074734.
International Preliminary Report on Patentability and English translation thereof dated May 14, 2015 in connection with Application No. PCT/JP2013/074744.
International Search Report and English translation thereof dated Feb. 13, 2018 in connection with International Application No. PCT/JP2017/044374.
International Search Report and Written Opinion and English translation thereof dated Oct. 18, 2013 in connection with Application No. PCT/JP2013/074734.
International Search Report and Written Opinion and English translation thereof dated Oct. 8, 2013 in connection with Application No. PCT/JP2013/074744.
Japanese Office Action dated May 16, 2017 in connection with Japanese Application No. 2014-544375 and English translation thereof.
Partial Supplementary European Search Report dated Jun. 14, 2016 in connection with European Application No. 13850571.4.
U.S. Appl. No. 14/437,884, filed Apr. 23, 20215, Asada.
U.S. Appl. No. 14/438,437, filed Apr. 24, 2015, Asada et al.
U.S. Appl. No. 16/201,794, filed Nov. 27, 2018, Asada et al.
Written Opinion and English translation thereof dated Feb. 13, 2018 in connection with International Application No. PCT/JP2017/044374.

Also Published As

Publication numberPublication date
CN110226200A (en)2019-09-10
KR20190113778A (en)2019-10-08
JP7020432B2 (en)2022-02-16
EP3579225A4 (en)2020-02-19
EP3579225A1 (en)2019-12-11
US20190385585A1 (en)2019-12-19
WO2018142770A1 (en)2018-08-09
JPWO2018142770A1 (en)2019-11-21

Similar Documents

PublicationPublication DateTitle
US8483396B2 (en)Method for the sound processing of a stereophonic signal inside a motor vehicle and motor vehicle implementing said method
US9245517B2 (en)Noise reduction audio reproducing device and noise reduction audio reproducing method
US9236041B2 (en)Filter circuit for noise cancellation, noise reduction signal production method and noise canceling system
JP4509686B2 (en) Acoustic signal processing method and apparatus
JP2016510915A5 (en)
WO2013077226A1 (en)Audio signal processing device, audio signal processing method, program, and recording medium
JP6870078B2 (en) Noise estimation for dynamic sound adjustment
US10896668B2 (en)Signal processing apparatus, signal processing method, and computer program
EP3061268A1 (en)Method and mobile device for processing an audio signal
JP4967894B2 (en) Signal processing apparatus, signal processing method, program, noise canceling system
EP2337020A1 (en)A device for and a method of processing an acoustic signal
US20070223750A1 (en)Crosstalk cancellation system with sound quality preservation and parameter determining method thereof
JP6197930B2 (en) Ear hole mounting type sound collecting device, signal processing device, and sound collecting method
US10681487B2 (en)Acoustic signal processing apparatus, acoustic signal processing method and program
WO2016059878A1 (en)Signal processing device, signal processing method, and computer program
JP2015211418A (en)Acoustic signal processing device, acoustic signal processing method and program
CN111492669B (en)Crosstalk cancellation for oppositely facing earspeaker systems
TW202018700A (en)Spatial crosstalk processing for stereo signal
TW201909656A (en) Compensation for crosstalk and subband spatial processing
US20070081674A1 (en)Method and apparatus of amplifying stereo effect
JP2006217210A (en) Audio equipment
JP2008048324A (en)Automatic panning adjusting apparatus and method
JP4402636B2 (en) Audio equipment
JP4402632B2 (en) Audio equipment
US11689872B2 (en)Acoustic device with first sound outputting device for input signal, second outputting device for monaural signal and L-channel stereo component and third sound outputting device for monaural signal and R-channel stereo component

Legal Events

DateCodeTitleDescription
FEPPFee payment procedure

Free format text:ENTITY STATUS SET TO UNDISCOUNTED (ORIGINAL EVENT CODE: BIG.); ENTITY STATUS OF PATENT OWNER: LARGE ENTITY

ASAssignment

Owner name:SONY CORPORATION, JAPAN

Free format text:ASSIGNMENT OF ASSIGNORS INTEREST;ASSIGNORS:TAMORI, YOSHINORI;ASADA, KOHEI;ITABASHI, TETSUNORI;AND OTHERS;SIGNING DATES FROM 20190605 TO 20190606;REEL/FRAME:050050/0814

STPPInformation on status: patent application and granting procedure in general

Free format text:FINAL REJECTION MAILED

STPPInformation on status: patent application and granting procedure in general

Free format text:DOCKETED NEW CASE - READY FOR EXAMINATION

STPPInformation on status: patent application and granting procedure in general

Free format text:PUBLICATIONS -- ISSUE FEE PAYMENT VERIFIED

STCFInformation on status: patent grant

Free format text:PATENTED CASE

MAFPMaintenance fee payment

Free format text:PAYMENT OF MAINTENANCE FEE, 4TH YEAR, LARGE ENTITY (ORIGINAL EVENT CODE: M1551); ENTITY STATUS OF PATENT OWNER: LARGE ENTITY

Year of fee payment:4


[8]ページ先頭

©2009-2025 Movatter.jp